| // Copyright (c) 2012 The Chromium Authors. All rights reserved. |
| // Use of this source code is governed by a BSD-style license that can be |
| // found in the LICENSE file. |
| |
| #include "content/renderer/media/webaudio_capturer_source.h" |
| |
| #include "base/logging.h" |
| #include "base/time/time.h" |
| #include "content/renderer/media/webrtc_audio_capturer.h" |
| #include "content/renderer/media/webrtc_local_audio_track.h" |
| |
| using media::AudioBus; |
| using media::AudioFifo; |
| using media::AudioParameters; |
| using media::ChannelLayout; |
| using media::CHANNEL_LAYOUT_MONO; |
| using media::CHANNEL_LAYOUT_STEREO; |
| |
| static const int kMaxNumberOfBuffersInFifo = 5; |
| |
| namespace content { |
| |
| WebAudioCapturerSource::WebAudioCapturerSource() |
| : track_(NULL), |
| capturer_(NULL), |
| audio_format_changed_(false) { |
| } |
| |
| WebAudioCapturerSource::~WebAudioCapturerSource() { |
| } |
| |
| void WebAudioCapturerSource::setFormat( |
| size_t number_of_channels, float sample_rate) { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| DVLOG(1) << "WebAudioCapturerSource::setFormat(sample_rate=" |
| << sample_rate << ")"; |
| if (number_of_channels > 2) { |
| // TODO(xians): Handle more than just the mono and stereo cases. |
| LOG(WARNING) << "WebAudioCapturerSource::setFormat() : unhandled format."; |
| return; |
| } |
| |
| ChannelLayout channel_layout = |
| number_of_channels == 1 ? CHANNEL_LAYOUT_MONO : CHANNEL_LAYOUT_STEREO; |
| |
| base::AutoLock auto_lock(lock_); |
| // Set the format used by this WebAudioCapturerSource. We are using 10ms data |
| // as buffer size since that is the native buffer size of WebRtc packet |
| // running on. |
| params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
| channel_layout, number_of_channels, sample_rate, 16, |
| sample_rate / 100); |
| audio_format_changed_ = true; |
| |
| wrapper_bus_ = AudioBus::CreateWrapper(params_.channels()); |
| capture_bus_ = AudioBus::Create(params_); |
| audio_data_.reset( |
| new int16[params_.frames_per_buffer() * params_.channels()]); |
| fifo_.reset(new AudioFifo( |
| params_.channels(), |
| kMaxNumberOfBuffersInFifo * params_.frames_per_buffer())); |
| } |
| |
| void WebAudioCapturerSource::Start( |
| WebRtcLocalAudioTrack* track, WebRtcAudioCapturer* capturer) { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| DCHECK(track); |
| base::AutoLock auto_lock(lock_); |
| track_ = track; |
| capturer_ = capturer; |
| } |
| |
| void WebAudioCapturerSource::Stop() { |
| DCHECK(thread_checker_.CalledOnValidThread()); |
| base::AutoLock auto_lock(lock_); |
| track_ = NULL; |
| capturer_ = NULL; |
| } |
| |
| void WebAudioCapturerSource::consumeAudio( |
| const blink::WebVector<const float*>& audio_data, |
| size_t number_of_frames) { |
| base::AutoLock auto_lock(lock_); |
| if (!track_) |
| return; |
| |
| // Update the downstream client if the audio format has been changed. |
| if (audio_format_changed_) { |
| track_->OnSetFormat(params_); |
| audio_format_changed_ = false; |
| } |
| |
| wrapper_bus_->set_frames(number_of_frames); |
| |
| // Make sure WebKit is honoring what it told us up front |
| // about the channels. |
| DCHECK_EQ(params_.channels(), static_cast<int>(audio_data.size())); |
| |
| for (size_t i = 0; i < audio_data.size(); ++i) |
| wrapper_bus_->SetChannelData(i, const_cast<float*>(audio_data[i])); |
| |
| // Handle mismatch between WebAudio buffer-size and WebRTC. |
| int available = fifo_->max_frames() - fifo_->frames(); |
| if (available < static_cast<int>(number_of_frames)) { |
| NOTREACHED() << "WebAudioCapturerSource::Consume() : FIFO overrun."; |
| return; |
| } |
| |
| fifo_->Push(wrapper_bus_.get()); |
| int capture_frames = params_.frames_per_buffer(); |
| base::TimeDelta delay; |
| int volume = 0; |
| bool key_pressed = false; |
| if (capturer_) { |
| capturer_->GetAudioProcessingParams(&delay, &volume, &key_pressed); |
| } |
| |
| // Turn off audio processing if the delay value is 0, since in such case, |
| // it indicates the data is not from microphone. |
| // TODO(xians): remove the flag when supporting one APM per audio track. |
| // See crbug/264611 for details. |
| bool need_audio_processing = (delay.InMilliseconds() != 0); |
| while (fifo_->frames() >= capture_frames) { |
| fifo_->Consume(capture_bus_.get(), 0, capture_frames); |
| // TODO(xians): Avoid this interleave/deinterleave operation. |
| capture_bus_->ToInterleaved(capture_bus_->frames(), |
| params_.bits_per_sample() / 8, |
| audio_data_.get()); |
| track_->Capture(audio_data_.get(), delay, volume, key_pressed, |
| need_audio_processing, false); |
| } |
| } |
| |
| } // namespace content |